The channel bandwidth of a wireless communication system is much narrower than that of a conventional telephone communication system of 64 kbps, and thus digital audio data in a wireless communication system is compressed before being transmitted. Methods for compressing digital audio data in a wireless communication system include QCELP (QualComm Code Excited Linear Prediction) of IS-95, EVRC (Enhanced Variable Rate Coding), VSELP (Vector-Sum Excited Linear Prediction) of GSM (Global System for Mobile Communication), RPE-LTP (Regular-Pulse Excited LPC with a Long-Term Predictor), and ACELP (Algebraic Code Excited Linear Prediction). All of these listed methods are based on LPC (Linear Predictive Coding). Audio compressing methods based on LPC utilize a model optimized to human voices and thus are efficient to compress voice at a low or middle encoding rate. In a coding method used in a wireless system, to efficiently use the limited bandwidth and to decrease power consumption, digital audio data is compressed and transmitted only when speaker's voice is detected by using what is called the function of VAD (Voice Activity Detection).
There are various reasons why the perceptual sound quality of digital audio data is degraded after the digital audio data is compressed using audio codecs based on LPC, especially EVRC codecs. The perceptual sound quality degradation occurs in the following ways.                (i) Complete loss of frequency components in a high-frequency bandwidth        (ii) Partial loss of frequency components in a low-frequency bandwidth        (iii) Intermittent pause of music        
The first cause of the degradation cannot be avoided as long as the high-frequency components are removed using a 4 kHz (or 3.4 kHz) lowpass filter when digital audio data is compressed using narrow bandwidth audio codec.
The second phenomenon is due to the intrinsic characteristic of the audio compression method based on LPC. According to the LPC-based compression methods, a pitch and a formant frequency of an input signal are obtained, and then an excitation signal for minimizing the difference between the input signal and the composite signal calculated by the pitch and the formant frequency of the input signal, is derived from a codebook. It is difficult to extract a pitch from a polyphonic music signal, whereas it is easy in case of a human voice. In addition, the formant component of music is very different from that of a person's voice. Consequently, it is expected that the prediction residual signals for music data would be much larger than those of human speech signal, and thus many frequency components included in the original digital audio data are lost. The above two problems, that is, loss of high and low frequency components are due to inherent characteristic of audio codecs optimized to voice signals, and inevitable to a certain degree.
The pauses in digital audio data are caused by the variable encoding rate used by EVRC. An EVRC encoder processes the digital audio data with three rates (namely, 1, ½, and ⅛). Among these rates, ⅛ rate means that the EVRC encoder determines that the input signal is a noise, and not a voice signal. Because sound of a percussion instrument, such as a drum, include spectrum components that tend to be perceived as noises by audio codecs, music including this type of sound is frequently paused. Also, audio codecs consider sound having a low amplitude as noises, which also degrade the perceptual sound quality.
Recently, several services for providing music to wireless phone users became available. One of which is what is called “Coloring service” which enables a subscriber to designate a tune of his/her choice so that callers who make a call to the subscriber would hear music instead of the traditional ringing tone until the subscriber answers the phone. Since this service became very popular first in Korea where it originated and then in other countries, transmission of music data to a cellular phone has been increasing. However, as explained above, the audio compression method based on LPC is suitable for human voice that has limited frequency components. When music or signals having frequency components spread out through the audible frequency range (20-20,000 Hz) are processed in a conventional LPC based codecs and transmitted through a cellular system, signal distortion occurs, which causes pauses in music.